On Thu, Apr 20, 2017 at 05:51:59PM -0300, Fabio Moretti wrote: > Il 20/04/2017 17:32, kevin.lar...@pioneerballoon.com ha scritto: > > > > This gets kinda Rube Golberg-ish, but convert the incoming analog line > > to sip, route it through asterisk and have asterisk do its thing > > before converting it back to analog to send to the phone. Only problem > > is you get a lot of extra hardware involved in the mix to make it > > work. It will be a lot of expense and trouble, so you need to make > > sure that whatever part you want asterisk to play is worth that > > effort. Also, I wouldn't touch a fax line in this manner. > > > > If you could give a bit more info on what you want asterisk to do, we > > could maybe give better advice on how to solve your problem. > > Hi Kevin, > > I've already proposed your solution (is the most reasonable) but they > have more than 60 analogs lines (no faxes) and some of them terminate in > appliances like alarms, etc, so the solution must not touch in any way > the connection between the line and his termination: doing a analog to > digital conversion, passing it to asterisk and the convert it back to > analog is prone to problems (what if asterisk crashes? or if a gateway > fail?). > I can split the existing lines (there are no complex things like adsl or > digital signaling), convert the branches to digital and terminate then > into an asterisk machine, so any failure will not affect the old > circuit, but of course I've to configure asterisk to ONLY LOG calls and > nothing more. > > This is what they want: > - line 1 ring > - line 1 is splitted in two, the first branch (let's say the "analog" > branch) go to an analog phone, that rings > - the second branch go through a gateway and then to asterisk > - asterisk log (with an AGI for example) "line 1 rings at .... from ...."
Simple dialplan. Depending on the type of caller ID system, you may need to wait a few seconds (in case the caller ID is sent after the first ring). Thus, assuming you have a DAHDI device, your dialplan is: exten => s,1,Wait(5) ; check how much and if waiting is needed same => s,n,NoOp(Caller ID is ${CALLERID(num)} on DAHDI channel ${CHANNEL(dahdi_channel)}) And move on to report from there. If you also need to report the total time of the call: that might be possible if the remote side reverses polarity of the channels on call start and end. Information about it is currently only reported in debug messages by chan_dahdi. So it is possible (given polarity reversal), but tricky. -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users